Combined frequency response and dynamic range correction for loudspeakers

ABSTRACT

An audio processing apparatus includes a digital filter, a signal combiner, and a dynamic-range processor (DRP). The digital filter is configured to receive a filter-input audio signal and to filter the filter-input audio signal, so as to produce a filter-output audio signal. The signal combiner is configured to combine the filter-input audio signal with the filter-output audio signal, so as to produce a combined audio signal. The DRP is configured to apply to the filter-output audio signal a dynamic-range correction that depends non-linearly on the combined audio signal.

FIELD OF THE INVENTION

The present invention relates generally to processing of audio signals,and particularly to methods, systems and software for correction ofaudio signal distortions.

BACKGROUND OF THE INVENTION

Techniques for correcting audio signal distortions have been previouslyproposed in the patent literature. For example, U.S. Pat. No. 7,194,096describe a method in which an input audio signal is analyzed todetermine a power spectral density profile. The power spectral densityprofile is compared with at least one template profile. On the basis ofthe comparison, frequency bands of the input audio signal areselectively attenuated.

As another example, Russian patent RU 2,284,648 describes a method ofautomatic adaptive frequency correction of an audio signal. The methodis characterized by forming a reference spectrum, determining thespectrum of the source signal, comparing the spectrum of the sourcesignal and the reference spectrum, generating comparison signalsdepending on the comparison results, and then, using comparison signals,acting on the original signal for changes in the relations between itsspectral components to correct a signal.

U.S. Pat. No. 8,467,547 describes an audio compressor that may regulatethe level of an audio input signal depending on whether the level isabove or below a threshold value. The audio compressor may control apumping that is created when regulating a dynamic-range of an audiosignal with the threshold value. A feedback loop connecting the signaloutput from the audio compressor with a release filter may be used tomodify an effective release time of the signal. A controller may be usedthat allows a filter coefficient of the release filter to be controlledto adjust the effective release time as a function of the signal output.

SUMMARY OF THE INVENTION

An embodiment of the present invention provides an audio processingapparatus including a digital filter, a signal combiner, and adynamic-range processor (DRP). The digital filter is configured toreceive a filter-input audio signal and to filter the filter-input audiosignal, so as to produce a filter-output audio signal. The signalcombiner is configured to combine the filter-input audio signal with thefilter-output audio signal, so as to produce a combined audio signal.The DRP is configured to apply to the filter-output audio signal adynamic-range correction that depends non-linearly on the combined audiosignal.

In some embodiments, the digital filter is configured to both (i)equalize the filter-input audio signal and (ii) apply a partialcorrection for a resonance effect occurring in a loudspeaker driven bythe filter-output audio signal corrected by the DRP.

In some embodiments, the DRP is configured to apply a complementarycorrection to the resonance effect, in addition to the partialcorrection applied by the digital filter.

In an embodiment, prior to combining the filter-input audio signal withthe filter-output audio signal, the signal combiner is configured toapply a first weight to the filter-input audio signal and a secondweight to the filter-output audio signal.

In another embodiment, one or both of the first and second weights areuser-configurable.

In some embodiments, in at least one operational mode of the signalcombiner, the first weight is 1 and the second weight is −1. In otherembodiments, the first weight is selectable from the set {0,1} and thesecond weight is selectable from the set {−1,0,1}.

In an embodiment, the DRP is configured to apply a non-zerodynamic-range correction only when a power level of the combined audiosignal exceeds a threshold.

In some embodiments, the DRP includes one of a dynamic-range compressor(DRC), a limiter and an expander. There is additionally provided, inaccordance with another embodiment, an audio processing method includingreceiving in a digital filter a filter-input audio signal and to filterthe filter-input audio signal, so as to produce a filter-output audiosignal. The filter-input audio signal is combined with the filter-outputaudio signal in a signal combiner, so as to produce a combined audiosignal. Using a dynamic-range processor (DRP), a dynamic-rangecorrection that depends non-linearly on the combined audio signal isapplied to the filter-output audio signal.

The present invention will be more fully understood from the followingdetailed description of the embodiments thereof, taken together with thedrawings in which:

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a block diagram schematically illustrating an audio processingapparatus comprising a configurable digital filter mimic (DF-mimic) anda dynamic-range processor (DRP), in accordance with an embodiment of thepresent invention;

FIG. 2 is a state table for a signal combiner comprised in theconfigurable digital filter mimic (DF-mimic) of FIG. 1, in accordancewith an embodiment of the present invention;

FIG. 3 is a graph of filter-input signal characteristic andfilter-output signal characteristic, as well as of an actual combinedsignal, in accordance with an embodiment of the present invention;

FIG. 4 is a graph of the output signal of the apparatus of FIG. 1 bothwith and without resonance correction using the combined signal of FIG.3, in accordance with an embodiment of the present invention;

FIG. 5 is a graph of the output signal of the apparatus of FIG. 1, bothwith and without a resonance correction, in accordance with anotherembodiment of the present invention; and

FIG. 6 is a flow chart that schematically illustrates a method forcorrecting an audio output signal using the audio processing apparatusof FIG. 1, in accordance with an embodiment of the present invention.

DETAILED DESCRIPTION OF EMBODIMENTS Overview

Manufacturers of consumer-grade loudspeakers, in particular those fittedin mobile devices, face the challenge of balancing sufficient audioquality with price. Particularly challenging are small plasticloudspeaker enclosures, common in the consumer electronics market, whichare prone to suffer from audio resonance phenomena. For example, acommon audio quality problem may occur when a loudspeaker box, such asfitted inside, e.g., a mobile phone, undesirably “resonates” at aspecific frequency. In such a case the outputted acoustic energy at theresonance frequency is significantly higher than expected, and alsohigher than on other frequencies. Such an effect is commonly thought tooccur due to air pressure resonance created within the loudspeaker'senclosure and is, as mentioned above, an unwanted audio artifact.

Moreover, such resonance effects (e.g., standing waves in an enclosure)can build up in a non-linear way. For example, a common problem in theconsumer electronics field is for a loudspeaker to produce lessnoticeable audio resonance when input signal amplitudes are low (e.g.,producing a low audio amplitude “peak” below a threshold value), andhaving the amplitude of the unwanted audio resonant artifact growingnon-linearly with input signal amplitude, in the range above thethreshold value.

One possible way to solve the above audio frequency and power relatedproblems is by digital signal-processing (DSP) techniques, for exampleby applying to the input signal a digital-filter (DF) and adynamic-range processor (DRP). The DRP may include such options as adynamic-range compressor (DRC), a limiter or an expander, among others.However, this solution is often too expensive to be fully implemented,as such dedicated DF and DRP combinations increase complexity as well ascomputational and electrical-power consumption. Moreover, when takinginto account that audio resonance may occur at multiple frequencies, aswell as the need to solve other audio problems, in many cases it isimpractical to include a sufficient number of additional dedicated DFand DRC pairs in a consumer device, to solve every resonance problem.

Embodiments of the present invention that are described hereinafterprovide an apparatus that utilizes one or more digital filters alreadycomprised in the consumer device (e.g., in an equalizer unit of thedevice) to also function, in addition to their original uses, as DFsthat provide DF inputs to DRPs. To facilitate this dual use, thedisclosed technique combines the filter-input audio signal with thefilter-output audio signal, using a signal combiner. The combined signalis provided as input to the DRP. Acting together, the digital filter andthe signal combiner are collectively named hereinafter “configurableDF-mimic.” As the signal combiner includes substantially fewer elementsthan a dedicated DF, and considering the multiple dedicated DFs thatmight otherwise have been required, the embodiments of the disclosedinvention can save significant hardware and software resources in thedevice.

Moreover, the disclosed DF-mimic technique extends the possibilities ofthe apparatus to handle other audio-quality issues, by providing, e.g.,using the equalizer's digital filter, a configurable equivalent of theDF, thereby implementing a configurable signal correction circuitry(i.e., using combined configurable-DF-mimic and DRP). As an equalizermay include multiple digital filters, the disclosed technique enablesmultiple configurable signal correction circuitry (i.e., combinations ofDF mimics and respective DRPs) of this type using already existingfilters.

As noted above, a key ingredient of the disclosed technique is tocombine the filter input and filter output signals. Therefore, in someembodiments, the disclosed audio processing apparatus includes (a) theaforementioned digital filter (typically a linear DF) comprised in thedevice (e.g., a linear DF comprised in an equalizer of the device),wherein the DF is configured to receive a filter-input audio signal andto filter the filter-input audio signal so as to produce a filter-outputaudio signal, (b) a signal combiner, configured to combine thefilter-input audio signal with the filter-output audio signal, so as toproduce a combined audio signal of the above-described DF-mimic, and (c)an existing DRP (e.g., a DRC) configured to apply a dynamic-rangecorrection to the filter-output audio signal that depends non-linearlyon the combined audio signal.

In one embodiment, the above-mentioned digital filter is configured toapply a partial correction to a resonance effect occurring in aloudspeaker driven by the filter-output audio signal. The signalcombiner is configured to apply a first weight to the filter-input audiosignal and a second weight to the filter-output audio signal and outputthe combined signal to the DRP. The DRP is then configured to apply acomplementary correction to the resonance effect, in addition to thepartial correction applied by the digital filter.

As further noted above, the disclosed technique is configurable and, inanother embodiment, one or both of the first and second weights areconfigurable per device model. For example, the disclosed solution maybe provided with a state table for the configurable DF-mimic, in whichthe first and second weights are selected to form differentcombinations. The different possible combinations greatly expand thecapabilities of a signal correction circuitry to perform DSP operationsto overcome loudspeaker imperfections, as well as backward-compatibilityand testing capabilities. For example, in one embodiment, the disclosedconfigurable DF-mimic is configured to serve as a band-pass filter(BPF). In another example, the disclosed configurable DF-mimic isconfigured to serve as a band-stop filter (BSF).

By providing audio control with combined DF-mimic and DRP solutionscomprising configurable DF-mimics using available digital-filters in adevice, the disclosed technique fills the requirements of providingimproved audio quality control in consumer-grade devices whilemaintaining low complexity, and low computational and electrical-energyrequirements.

Combined Frequency Response and Dynamic-Range Corrector

for Loudspeaker Correction

As noted above, in some consumer devices, a spectrally-uniform inputaudio signal may nevertheless generate a loudspeaker resonance at acertain frequency. Had the resonance amplitude been a linear function ofthe input signal amplitude, the audio resonance could have been morereadily mitigated by using a digital filter. For example, assuming arelative peak resonance of S decibels, a digital filter having afrequency response comprising a (−S) [dB] notch at the resonantfrequency would have resolved the problem by canceling the audioresonance.

In many practical cases, however, S is not a fixed value, since theresonance tends to be weak at low input signal amplitudes and to grow asinput signal amplitude increases. To compensate for such a dependence ofS, a combination of a DF and DRP functionally is required, which,without the disclosed embodiments of the invention, require an apparatuscomprising a dedicated DF on an input path to a DRP.

FIG. 1 is a block diagram schematically illustrating an audio-processingapparatus 10 comprising a signal correction circuitry 25 comprisingconfigurable digital filter mimic (DF-mimic) 20 and a dynamic-rangeprocessor (DRP) 22, in accordance with an embodiment of the presentinvention. Processing apparatus 10 may be used in any suitable system ordevice that outputs audio signals, e.g., in a mobile phone, a computer,a gaming console or a stereo system, to name only a few possibilities.

Processing apparatus 10 has a feed-forward topology in which,responsively to an analyzed filter-input audio signal 111, adynamic-range correction (e.g., a corrective gain coefficient) iscalculated and applied (222) to generate an output audio signal 333 inwhich undesired resonance effects are suppressed considerably.

In more detail, signal correction circuitry 25 comprises a configurableDF-mimic module 20 that outputs a combined audio signal 220 for use asan input of a Dynamic-Range Corrector (DRP) 22. Responsively to thecombined signal, DRP 22 calculates a gain coefficient 222 that is theoutput of signal correction circuitry in 25. In the shown embodiment,configurable DF-mimic 20 is realized by a linear digital filter (LDF) 12and by a signal combiner 19 to implement the disclosed signal correctioncircuitry 25 functionality.

LDF 12, usually a second order filter, usually of the type “Direct Form1” (DF1) or “direct Form 2” (DF2) or similar, is shown in FIG. 1 as anon-limiting example. Typically, LDF 12—or any other filter type used—isone of a set of filters of an equalizer unit (not shown). LDF 12 plays adual role in apparatus 10—(i) Equalizes input audio signal 111, and (ii)applies partial correction for loudspeaker resonance. Other equalizerunit filters may also be used, e.g., in parallel to LDF 12, to solveother audio quality issues. For example, other available digital filtersmay be comprised in configurable DF-mimics to solve other aforementionedloudspeaker resonances occurring at other frequencies.

As noted above, in the shown embodiment, adding a dedicated DF isavoided since apparatus 10 comprises signal combiner 19 configured tocombine a filter-input audio signal 111 (also called “Dry Pre LDF”signal) with a filter-output audio signal 113 (also called “Wet PostLDF” signal), so as to produce a combined audio signal 220 of DF-mimic20. As seen, LDF 12 input signals and output signals are combined (e.g.,by adder 18), after the two signals are multiplied with respective firstand second weights G_(d) and G_(w), which are described in FIG. 2.

In response to combined audio signal 220 of DF-mimic 20, DRP 22 isconfigured to apply (222) a dynamic-range correction to filter-outputaudio signal 113 that depends non-linearly on combined audio signal 220.The resulting audio output, as shown in FIG. 4, is essentially the sameas if a dedicated BPF had been added to signal correction circuitry 25.Since a dedicated BPF includes significantly more elements than signalcombiner 19, and since multiple dedicated BPFs may have been needed in asingle device, the disclosed technique can save significant hardware andsoftware resources.

The example embodiment of FIG. 1 is depicted by way of example, and in asimplified way, for the sake of clarity. For example, other types ofdigital filters may be used by the disclosed technique, such as asingle-pole filter, a three-pole or any other number of poles filter,LPF, BPF of any order, finite impulse response (FIR) filters, etc.Additional elements of apparatus 10, such as other components of theequalizer unit, and audio amplification stages, are not described forclarity of presentation.

In various embodiments, the different elements of the audio processingapparatus shown in FIG. 1 may be implemented using suitable hardware,such as using one or more discrete components, one or moreApplication-Specific Integrated Circuits (ASICs) and/or one or moreField-Programmable Gate Arrays (FPGAs). Some of the functions of thedisclosed audio processing apparatuses, e.g., some or all functions ofDF-mimic 20 and DRP 22, may be implemented in one or more generalpurpose processors, which are programmed in software to carry out thefunctions described herein. The software may be downloaded to theprocessors in electronic form, over a network or from a host, forexample, or it may, alternatively or additionally, be provided and/orstored on non-transitory tangible media, such as magnetic, optical, orelectronic memory.

FIG. 2 is a state table for the signal combiner 19 comprised inconfigurable digital filter mimic (DF-mimic) 20 of FIG. 1, in accordancewith an embodiment of the present invention. As seen, the state tablecomprises three modes (A, B, C) for configuring DF-mimic 20. The tableincludes first and second weights G_(d) and G_(w) values, that, by wayof example, are selected to be either zero or ±1.

Setting G_(d) to +1 and G_(w) to −1 (“mode B” of the table) generates aBPF-like response curve of DF-mimic 20, for proper selection of LDF 12,such as selecting direct-form 1 (DF1) or direct-form 2 (DF2) filters andsetting the selected filter to function as a notch filter. The BPFresponse achieved in “mode B” is typically the most desired shape forthe input of DRP 22. This is because the resulting BPF characteristics(which serves as input DRP 22) is particularly sensitive to the resonantfrequency, less to other frequencies.

However, the user may still want to use “mode A” in which weights G_(d)and G_(w) are set to have DF-mimic 20 outputs the same output as that ofLDF 12. This is the same design as in existing systems in which thelinear filter is placed before the DRP and can be used for backwardcompatibility if this solution is good enough.

Another option is using “mode C” in which weights G_(d) and G_(w) areset to have DF-mimic 20 provide a frequency-flat input to DRP 22. Thisis the same design as in existing systems in which the linear filter isplaced after the DRC and can be used for backward compatibility if thissolution is good enough.

With these three modes, the user can easily set up an existing apparatus10 to operate in one of three modes:

-   -   Mode B: sensitive to resonant frequency    -   Mode A: mimic a linear filter before the DRC (cascaded design)    -   Mode C: mimic the linear filter after the DRC (cascaded design)

These three modes are achieved with a minimum of components added to thedevice to form apparatus 10 (mainly to form signal combiner 19),yielding low computational resource requirements and high functionalflexibility.

In general, first and second weights G_(d) and G_(w) can have othervalues, and depending on the values of the first and second weightsG_(d) and G_(w), configurable DF-mimic 20 can be preset to have anycharacteristics between a strong BPF (e.g., after “mode B”) to a strongBSF (e.g., after “mode A”).

For the sake of clarity of presentation, the above treatment is givenmostly in general terms. As a complement, FIGS. 3 and 4 provide a fewspecific practical implementations, using non-trivial parameters, andfurther provide real-time results measured with a standard spectrumanalyzer to reflect the output of a real apparatus employing thedisclosed technique.

All results discussed in this paragraph were measured (in real-time) byan Audio Precision AP525 audio analyzer, accepted in this industry as astandard. The results below include both FR (frequency response) and DR(dynamic-range) measurements of two embodiments of the application,implemented in software on a Windows® 10 PC and run with real-time audiodesign tools.

FIG. 3 is a graph of filter-input signal characteristic (1111) andfilter-output signal characteristic (1131), as well as of an actualcombined signal 2201, in accordance with an embodiment of the presentinvention.

In FIG. 3 apparatus 10 is configured to correct an audio resonanceoccurring at a frequency of 1 [KHz], that a flat input signal 1111generates due to, for example, the aforementioned device enclosureimperfection. In the given example, there is a need to attenuate theoutput signal at a 1 [KHz] frequency region to be no more than (−20)[dB], without affecting other frequency regions.

Moreover, the specific frequency region of 1 [KHz] should be attenuatedby at least 10 [dB] along all of the dynamic-range of the input signal.In this non-limiting example, the resonant artifacts become more andmore dominant as the input goes over (−30) [dB].

To sum up the challenge, it requires the disclosed apparatus to limitthe system in this manner:

-   -   All frequencies except for 1 [KHz] region have no limit on FR or        DR    -   The amplitude at frequency region of 1 [KHz] should not pass        (−20) [dB] even if input is maximum (e.g., 0 [dB])    -   The frequency region of 1 [KHz] should have a linear attenuation        of (−10) [dB] as long as the input level (at around the        frequency 1 [KHz]) is lower than (−30) [dB]

Thus, apparatus 10 should reduce output amplitude 333 about the resonantfrequency only, and particularly, if the resonance amplitude becomes toohigh.

To achieve a combined signal 2201 seen in FIG. 3 that meets the aboverequirements, the apparatus is configured with a linear digital filterhaving a gain of (−10) [dB] at 1 [KHz], which, by selecting the Q-factorof the digital filter sufficiently high (in this case Q is set equal to12), produces a notch frequency response curve as seen by signal 1131.

DRC 22 can now be configured to attenuate a resonance by using theBPF-mimic frequency-response that DF-mimic 20 was configured to supply.In particular, DRC 22 is configured to output (222) a gain coefficientthat is used by multiplier 30 to further attenuate the already partiallyattenuated resonance audio resonance.

To achieve this result, the DRP parameters are set as follows: athreshold=(−30) [dB], and a ratio=1:1.5, where the DRP input/outputratio determines a compression ratio of signal intensity above thethreshold level (e.g., for a signal 0 [dB] signal, which is [30 dB]above the threshold, the ratio means the DRP will attenuates the signalby 20 [dB] on top of the 10 [dB] suppression provided by the filter).Other DRP settings, such as pre-gain, post-gain, attack, release, knee,etc. are not mentioned here in order to simplify the discussion.

FIG. 4 is a graph of the output signal of the apparatus of FIG. 1without (3330) resonance correction and with resonance correction (3331)using the combined signal 2201 of FIG. 3, in accordance with anembodiment of the present invention. As seen, output signal 3331 ofapparatus 10 at 1 [KHz] (after being corrected by a corrective gaincoefficient 222 outputted by DRP 22) has a knee at −30 [dB]. Compared tothe uncorrected signal 3330, output signal 3331 is 10 [dB] attenuatedfor input-signal amplitude lower than −30 [dB], and, as input-signalamplitude crosses the threshold, DRP 22 increasingly attenuates signal3331 to eliminate an otherwise increasing resonant amplitude of theloudspeaker signal (not shown) between −30 [dB] and 0 [dB] inputamplitude.

When inspecting frequencies other than resonance frequency (e.g.,inspecting audio quality at 10 [KHz]), the equivalent of signals 3330and 3333 are practically identical, as both the digital filter and theRDCR have no effect.

To demonstrate the flexibility of the disclosed apparatus, anotherimplementation of resonance suppression at 1 [Khz] is presented, inwhich the disclosed apparatus should not give more than a (−35) [dB]output signal. At the specific frequency region of 1 [KHz] the signalshould be attenuated by at least 15 [dB] along the entire dynamic-rangeof the input signal. In this additional non-limiting example, theresonant artifacts become more and more dominant as the input goes over(−40) [dB].

To achieve the above requirements, the apparatus is configured with alinear digital filter having a gain of (−15) [dB] at 1 [KHz] andQ-factor of 12. The DRP parameters are set as threshold=(−40) [dB], andratio=1:2.

Accordingly, FIG. 5 is a graph of the output signal of apparatus 10 ofFIG. 1, without (4330) and with (4331) the resonance correction, inaccordance with another embodiment of the present invention.

As seen in FIG. 5, output signal 4331 of apparatus 10 at 1 [KHz] (afterbeing corrected by a corrective gain coefficient 222 outputted by DRP22) has a knee at −40 [dB] input-signal amplitude. Compared with theuncorrected signal 4330, output signal 4331 is 15 [dB] attenuated forinput-signal amplitude lower than −40 [dB], and, as input-signalamplitude crosses the threshold, DRP 22 increasingly attenuates signal4331, to eliminate an otherwise increasing resonant amplitude of theloudspeaker signal (not shown) between −40 [dB] and 0 [dB] inputamplitude.

FIG. 6 is a flow chart that schematically illustrates a method forcorrecting an audio output signal using the audio processing apparatusof FIG. 1, in accordance with an embodiment of the present invention.The algorithm, according to the presented embodiment, carries out aprocess that begins with filter 12 of apparatus 10 filtering afilter-input audio signal 111 to produce a filter-output signal 113, ata digital filtration step 60.

At a signal weighting step 62, signal combiner 19 applies a firstweight, G_(d), to the filter-input audio signal and applies a secondweight, G_(w), to the filter-output audio signal. Next, at a signalcombining step 64, signal combiner 19 combines the weighted signalsusing adder 18.

At DRP inputting step 66, adder 18 outputs the combined signal (120)into an input port of DRP 22, which, at a corrective gain applicationstep 68, applies, using multiplier 30, a dynamic-range correction signal(222) to the filter-output audio signal 113 that depends on the combinedaudio signal, to suppress an audio resonance.

Although the embodiments described herein mainly address audioprocessing for consumer grade devices, the methods and systems describedherein can also be used in other applications, such as in musicinstruments with internal loudspeakers, such as digital pianos, and inportable audio (PA) systems, amplified speakers for pro-audio (studiomonitors), etc.

It will thus be appreciated that the embodiments described above arecited by way of example, and that the present invention is not limitedto what has been particularly shown and described hereinabove. Rather,the scope of the present invention includes both combinations andsub-combinations of the various features described hereinabove, as wellas variations and modifications thereof which would occur to personsskilled in the art upon reading the foregoing description and which arenot disclosed in the prior art. Documents incorporated by reference inthe present patent application are to be considered an integral part ofthe application except that to the extent any terms are defined in theseincorporated documents in a manner that conflicts with the definitionsmade explicitly or implicitly in the present specification, only thedefinitions in the present specification should be considered.

The invention claimed is:
 1. An audio processing apparatus, comprising:a digital filter, configured to receive a filter-input audio signal andto filter the filter-input audio signal, so as to produce afilter-output audio signal; a signal combiner, configured to combine thefilter-input audio signal with the filter-output audio signal, so as toproduce a combined audio signal; and a dynamic-range processor (DRP),configured to apply to the filter-output audio signal a dynamic-rangecorrection that depends non-linearly on the combined audio signal. 2.The apparatus according to claim 1, wherein the digital filter isconfigured to both (i) equalize the filter-input audio signal and (ii)apply a partial correction for a resonance effect occurring in aloudspeaker driven by the filter-output audio signal corrected by theDRP.
 3. The apparatus according to claim 2, wherein the DRP isconfigured to apply a complementary correction to the resonance effect,in addition to the partial correction applied by the digital filter. 4.The apparatus according to claim 1, wherein, prior to combining thefilter-input audio signal with the filter-output audio signal, thesignal combiner is configured to apply a first weight to thefilter-input audio signal and a second weight to the filter-output audiosignal.
 5. The apparatus according to claim 4, wherein one or both ofthe first and second weights are user-configurable.
 6. The apparatusaccording to claim 4, wherein, in at least one operational mode of thesignal combiner, the first weight is 1 and the second weight is −1. 7.The apparatus according to claim 4, wherein the first weight isselectable from the set {0,1} and the second weight is selectable fromthe set {−1,0,1}.
 8. The apparatus according to claim 1, wherein the DRPis configured to apply a non-zero dynamic-range correction only when apower level of the combined audio signal exceeds a threshold.
 9. Theapparatus according to claim 1, wherein the DRP comprises one of adynamic-range compressor (DRC), a limiter and an expander.
 10. An audioprocessing method, comprising: filtering a filter-input audio signal, soas to produce a filter-output audio signal; combining the filter-inputaudio signal with the filter-output audio signal, so as to produce acombined audio signal; and applying to the filter-output audio signal adynamic-range correction that depends non-linearly on the combined audiosignal.
 11. The method according to claim 10, wherein filtering thefilter-input audio signal comprises both (i) equalizing the filter-inputaudio signal and (ii) applying a partial correction for a resonanceeffect occurring in a loudspeaker driven by the dynamic-range correctedfilter-output audio signal.
 12. The method according to claim 11,wherein applying the dynamic-range correction comprises applying acomplementary correction to the resonance effect, in addition to thepartial correction applied in filtering the filter-input audio signal.13. The method according to claim 10, and comprising, prior to combiningthe filter-input audio signal with the filter-output audio signal,applying a first weight to the filter-input audio signal and a secondweight to the filter-output audio signal.
 14. The method according toclaim 13, wherein one or both of the first and second weights areuser-configurable.
 15. The method according to claim 13, wherein, in atleast one operational mode of the signal combiner, the first weight is 1and the second weight is −1.
 16. The method according to claim 13,wherein the first weight is selectable from the set {0,1} and the secondweight is selectable from the set {−1,0,1}.
 17. The method according toclaim 10, wherein applying the dynamic-range correction comprisesapplying a non-zero dynamic-range correction only when a power level ofthe combined audio signal exceeds a threshold.
 18. The method accordingto claim 10, wherein applying the dynamic-range correction comprisesapplying one of a dynamic-range compression, a dynamic-range limitationand a dynamic-range expansion.